Abstract:
A technology that can adaptively control an access mode according to a shared band or an unlicensed band used by an adjacent communication system. An access mode controlling apparatus includes a band recognition unit to recognize a usage rate of an unlicensed band or a shared band that is used by an adjacent communication system of a target communication system; a mode determination unit to determine, as an access mode, any one of a contention access mode and a contention-free access mode, based on the usage rate of the unlicensed band or the shared band; and an information transmitter to transmit mode information associated with the determined access mode to member nodes or a base station of the target communication system.
Abstract:
Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.
Abstract:
The present research relates to controlling rendering of multi-object or multi-channel audio signals. The present research provides a method and apparatus for controlling rendering of multi-object or multi-channel audio signals based on spatial cues in a process of decoding the multi-object or multi-channel audio signals. To achieve the purpose, the method suggested in the research controls rendering in a spatial cue domain in the process of decoding the multi-object or multi-channel audio signals.
Abstract:
A Unified Speech and Audio Codec (USAC) for adjusting an overlap area of a window based on a transition is provided. To increase an encoding efficiency, encoding may be performed by overlapping relatively long windows. Additionally, when a transition is generated between frames, an overlap area of a window may be reduced based on the transition, thereby preventing a noise from occurring due to the transition.
Abstract:
Provided are an apparatus and a method for reproducing a surround wave field using wave field synthesis. The apparatus includes an audio signal analyzer for analyzing a received multi-channel audio signal to check the number of audio signal channels, and extracting a sound source signal for each checked channel from the multi-channel audio signal; a wave field synthesis renderer for localizing the extracted sound source signal for each channel at a virtual sound image outside a narrow space using wave field synthesis so that the extracted sound source signal is suitable for the number of the checked audio signal channels; and an audio reproducer for reproducing the localized virtual sound source signal.
Abstract:
The present application discloses a system and a method for tag estimation and anti-collision in an RFID system, which can estimate an exact number of tags within an RF area and can rapidly identify tags by using the estimated number of tags in an RFID system. The system includes an RFID reader and RFID tags. The RFID reader includes an identification means, a collision management means, a tag number estimation means, and a reader control means. The RFID tag includes a tag communication means, a message reading means, a counter management means, an information storage means, and a tag control means. The disclosed system and method can prevent occurrence of too many idle times slots in the DFSA and too many initial collisions in the binary tree scheme, and thus can identify a large number of tags at a high speed with a small number of time slots.
Abstract:
Provided are an apparatus and method of separating, from a mixed signal, a sound source generated using a rhythm musical instrument based on characteristics of the rhythm musical instrument repeated in an aspect of time. The apparatus may include a separation unit to separate a plurality of mixed signals into a plurality of segments, a Nonnegative Matrix Partial Co-Factorization (NMPCF) analysis unit to perform an NMPCF analysis on the plurality of segments, and to obtain a plurality of entity matrices based on the analysis result, a target instrument signal separating unit to separate, from the mixed signals, a target instrument signal, by calculating an inner product between the plurality of entity matrices, and a signal association unit to associate the target instrument signals separated from each of the plurality of segments.
Abstract:
Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).
Abstract:
A module capable of appropriately selecting a linear predictive coding (LPC)-based or a code excitation linear prediction (CELP)-based speech or audio encoder and a transform-based audio encoder according to a feature of an input signal is a module that performs as a bridge for overcoming a performance barrier between a conventional LPC-based encoder and an audio encoder. Also, an integral audio encoder that provides consistent audio quality regardless of a type of the input audio signal can be designed based on the module.
Abstract:
Provided are an apparatus and a method for integrally encoding and decoding a speech signal and a audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is a audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; and a bitstream generator to generate a bitstream using an output signal of the first conversion encoder and an output signal of the LPC encoder.